A large-swing, high-driving, low-power, class-AB buffer amplifier, which consists of a high-gain input stage and a unity-gain class-AB output stage, with low variation of quiescent current is proposed. The low power consumption and low variation of the quiescent output current are achieved by using a weak-driving and a strong-driving pseudo-source followers. The high-driving capability is mainly provided by the strong-driving pseudo-source follower whose output transistors are turned off in the vicinity of the stable state to reduce the power consumption and the variation of output current, while the quiescent state is maintained by the weak-driving pseudo-source follower. The error amplifiers with source-coupled pairs of the same type transistors are merged into a single error amplifier to reduce the area of the buffer and the current consumption. An experimental prototype buffer amplifier implemented in a 0.35-µm CMOS technology demonstrates that the circuit dissipates an average static power consumption of only 388.7 µW with the standard deviation of only 3.4 µW, which is only 0.874% at a power supply of 3.3 V, and exhibits the slew rates of 2.18 V/µs and 2.50 V/µs for the rising and falling edges, respectively, under a 300 Ω /150 pF load. Both of the second and third harmonic distortions (HD2 and HD3) are -69 dB at 20 kHz under the same load.

A workflow definition containing errors might cause serious problems for an enterprise especially when it involves mission critical business processes or inter-organizational interaction. So workflow definitions should be defined in a strict and rigorous way. In this paper, we develop a workflow definition language and analysis methods for the language to support strict and rigorous workflow definitions. Faults or mistakes causing communication deadlock, access conflicts, and improper exception specification in workflow definitions can be detected and notified automatically using the methods. The proposed workflow definition language borrows structured constructs of conventional programming languages because many good features of conventional programming languages also can be used effectively in expressing workflow processes. With slight modifications and scope restrictions, the developed analysis techniques in this paper can be used in any workflow definition languages and they can help workflow designers define workflow processes in much more safe and reliable manner.

Optimistic coding is a coding in which we require the existence of reliable codes for infinitely many block length. In this letter we consider the optimistic source coding theorems for a general source Z from the information-spectrum approach. We first formulate the problem to be considered clearly. We obtain the optimistic infimum achievable source coding rate Tε (Z) for the case where decoding error probability εn is asymptotically less than or equal to an arbitrarily given ε [0,1). In fact, Tε (Z) turns out to be expressed in a form similar to the ordinary infimum achievable source coding rate. A new expression for Tε (Z) is also given. In addition, we investigate the case where εn = 0 for infinitely many n and obtain the infimum achievable coding rate.

The transformation of a data set using a second-order polynomial mapping to find statistically independent components is considered (quadratic independent component analysis or ICA). Based on overdetermined linear ICA, an algorithm together with separability conditions are given via linearization reduction. The linearization is achieved using a higher dimensional embedding defined by the linear parametrization of the monomials, which can also be applied for higher-order polynomials. The paper finishes with simulations for artificial data and natural images.

In this paper, the magnetic field properties around household appliances are investigated with the single coil model and equivalent source model, which are used as main source models in the European standard EN50366 (CENELEC). The accuracy of the field properties is conducted for the coil model (defined in the EN50366), by comparing with the results of the equivalent source model, which allow the reproduction of the complicated inhomogeneous magnetic field around the appliance with full generality (i.e. supports three dimensional vector fields).

In this letter, a resource allocation scheme is proposed to efficiently utilize the resource of CDMA systems with respect to the throughput. When we let the throughput be a function of various data rates for multimedia traffics, the scheme is to find the optimum set of data rates for concurrent user groups with which we can achieve the maximum throughput. It is also shown that the optimum data rate set always includes one non-trivial rate while all the others keep the minimum required data rate.

This paper addresses the utility-based radio resource allocation problem in DS-CDMA systems carrying multimedia traffic. The proposed scheme, aiming at achieving optimal resource allocation, considers the joint power and data rate allocation. To avoid high computational complexity of nonlinear optimization, we reformulate the radio resource allocation problem as a market model, where resource is regarded as a commodity. Since the market model satisfies the incentive-compatible constraint, the optimal resource allocation can be obtained at the market equilibrium in a distributed manner. According to whether to allocate a minimal transmission data rate to each user, two algorithms, UCA and FCA, are proposed. UCA emphasize on maximizing system overall utilities, while FCA guarantees fairness to users. Simulation results show that the proposed radio resource allocation scheme and algorithms are flexible and efficient for multimedia DS-CDMA systems.

A principal attraction of ATM networks, in both wired and wireless realizations, is that the key quality of service (QoS) parameters of every call, including end-to-end delay, jitter, and loss are guaranteed by the network when appropriate cell-level traffic controls are imposed at the user network interface (UNI) on a per call basis, utilizing the peak cell rate (PCR) and the sustainable cell rate (SCR) values for the multimedia--voice, video, and data, traffic sources. There are three practical difficulties with these guarantees. First, while PCR and SCR values are, in general, difficult to obtain for traffic sources, the typical user-provided parameter is a combination of the PCR, SCR, and the maximum burstiness over the entire duration of the traffic. Second, the difficulty in accurately defining PCR arises from the requirement that the smallest time interval must be specified over which the PCR is computed which, in the limit, will approach zero or the network's resolution of time. Third, the literature does not contain any reference to a scientific principle underlying these guarantees. Under these circumstances, the issue of providing QoS guarantees in the real world, through traffic controls applied on a per call basis, is rendered uncertain. This paper adopts a radically different, high level approach to the issue of QoS guarantees. It aims at uncovering through systematic experimentation a relationship, if any exists, between the key high level user traffic characteristics and the resulting QoS measures in a realistic operational environment. It may be observed that while each user is solely interested in the QoS of his/her own traffic, the network provider cares for two factors: (1) Maximize the link utilization in the network since links constitute a significant investment, and (2) ensure the QoS guarantees for every user traffic, thereby maintaining customer satisfaction. Based on the observations, this paper proposes a two-phase strategy. Under the first phase, the average "link utilization" computed over all the links in a network is maintained within a range, specified by the underlying network provider, through high level call admission control, i.e. by limiting the volume of the incident traffic on the network, at any time. The second phase is based on the hypothesis that the number of traffic sources, their nature--audio, video, or data, and the bandwidth distribution of the source traffic, admitted subject to a specific chosen value of "link utilization" in the network, will exert a unique influence on the cumulative delay distribution at the buffers of the representative nodes and, hence, on the QoS guarantees of each call. The underlying thinking is as follows. The cumulative buffer delay distribution, at any given node and at any time instant, will clearly reflect the cumulative effect of the traffic distributions of the multiple connections that are currently active on the input links. Any bounds imposed on the cumulative buffer delay distribution at the nodes of the network will also dominate the QoS bounds of each of the constituent user traffic. Thus, for each individual traffic source, the buffer delay distributions at the nodes of the network, obtained for different traffic distributions, may serve as its QoS measure. If the hypothesis is proven true, in essence, the number of traffic sources and their bandwidth distribution will serve asa practically realizable high level traffic control in providing realistic QoS guarantees for every call. To verify the correctness of the hypothesis, an experiment is designed that consists of a representative ATM network, traffic sources that are characterized through representative and realistic user-provided parameters, and a given set of input traffic volumes appropriate for a network provider approved link utilization measure. The key source traffic parameters include the number of sources that are incident on the network and the constituent links at any given time, the bandwidth requirement of the sources, and their nature. For each call, the constituent cells are generated stochastically, utilizing the typical user-provided parameter as an estimate of the bandwidth requirement. Extensive simulations reveal that, for a given link utilization level held uniform throughout the network, while the QoS metrics--end-to-end cell delay, jitter, and loss, are superior in the presence of many calls each with low bandwidth requirement, they are significantly worse when the network carries fewer calls of very high bandwidths. The findings demonstrate the feasibility of guaranteeing QoS for each and every call through high level traffic controls. As for practicality, call durations are relatively long, ranging from ms to even minutes, thereby enabling network management to exercise realistic controls over them, even in a geographically widely dispersed ATM network. In contrast, current traffic controls that act on ATM cells at the UNI face formidable challenge from high bandwidth traffic where cell lifetimes may be extremely short, in the range of µs. The findings also underscore two additional important contributions of this paper. First, the network provider may collect data on the high level user traffic characteristics, compute the corresponding average link utilization in the network, and measure the cumulative buffer delay distributions at the nodes, in an operational network. The provider may then determine, based on all relevant criteria, a range of input and system parameters over which the network may be permitted to operate, the intersection of all of which may yield a realistic network operating point (NOP). During subsequent operation of the network, the network provider may guide and maintain the network at a desired NOP by exercising control over the input and system parameters including link utilization, call admittance based on the requested bandwidth, etc. Second, the finding constitutes a vulnerability of ATM networks which a perpetrator may exploit to launch a performance attack.

We newly propose a novel blind separation framework for Single-Input Multiple-Output (SIMO)-model-based acoustic signals using an extended ICA algorithm, SIMO-ICA. The SIMO-ICA consists of multiple ICAs and a fidelity controller, and each ICA runs in parallel under the fidelity control of the entire separation system. The SIMO-ICA can separate the mixed signals, not into monaural source signals but into SIMO-model-based signals from independent sources as they are at the microphones. Thus, the separated signals of SIMO-ICA can maintain the spatial qualities of each sound source. In order to evaluate its effectiveness, separation experiments are carried out under both nonreverberant and reverberant conditions. The experimental results reveal that the signal separation performance of the proposed SIMO-ICA is the same as that of the conventional ICA-based method, and that the spatial quality of the separated sound in SIMO-ICA is remarkably superior to that of the conventional method, particularly for the fidelity of the sound reproduction.

This paper describes a real-time blind source separation (BSS) method for moving speech signals in a room. Our method employs frequency domain independent component analysis (ICA) using a blockwise batch algorithm in the first stage, and the separated signals are refined by postprocessing using crosstalk component estimation and non-stationary spectral subtraction in the second stage. The blockwise batch algorithm achieves better performance than an online algorithm when sources are fixed, and the postprocessing compensates for performance degradation caused by source movement. Experimental results using speech signals recorded in a real room show that the proposed method realizes robust real-time separation for moving sources. Our method is implemented on a standard PC and works in realtime.

We propose a new algorithm for overdetermined blind source separation (BSS) based on multistage independent component analysis (MSICA). To improve the separation performance, we have proposed MSICA in which frequency-domain ICA and time-domain ICA are cascaded. In the original MSICA, the specific mixing model, where the number of microphones is equal to that of sources, was assumed. However, additional microphones are required to achieve an improved separation performance under reverberant environments. This leads to alternative problems, e.g., a complication of the permutation problem. In order to solve them, we propose a new extended MSICA using subarray processing, where the number of microphones and that of sources are set to be the same in every subarray. The experimental results obtained under the real environment reveal that the separation performance of the proposed MSICA is improved as the number of microphones is increased.

Cluster computation has been used in the applications that demand performance, reliability, and availability, such as cluster server groups, large-scale scientific computations, distributed databases, distributed media-on-demand servers and search engines etc. In those applications, multicast can play the vital roles for the information dissemination among groups of servers and users. This paper proposes a set of novel efficient fault-tolerant multicast routing algorithms on hypercube interconnection of cluster computers using multicast shared tree approach. We present some new algorithms for selecting an optimal core (root) and constructing the shared tree so as to minimize the average delay for multicast messages. Simulation results indicate that our algorithms are efficient in the senses of short end-to-end average delay, load balance and less resource utilizations over hypercube cluster interconnection networks.

Differentiated Services architecture provides a framework that enables relative differentiation of Assured Forwarding (AF) service. The differentiation is quantified by QoS parameters in terms of loss probability and maximum delay. We develop herein an efficient model to compute resource allocation in terms of buffer and service rate that satisfies the QoS differentiation between classes of service. To evaluate the performance of the proposed model, we conducted extensive simulation on both single-node and multi-node cases. The simulation studies show that the model can provide an efficient method to allocate network resources for aggregated traffic.

A power and spreading gain allocation strategy is considered to effectively provide data services for mobile users with different levels of priorities in a DS-CDMA system. The priority level of a user is specified by a factor that is a weighting on the data throughput of the user. Our strategy implements a relative prioritization that affords a trade-off between spectral efficiency and strict prioritization.

In this paper, we analyzed the performance of dynamic resource allocation with channel de-allocation and buffering in cellular networks. Buffers are applied for data traffic to reduce the packet loss probability while channel de-allocation is exploited to reduce the voice blocking probability. The results show that while buffering data traffic can reduce the packet loss probability, it has negative impact on the voice performance even if channel de-allocation is exploited. Although the voice blocking probability can be reduced with large slot capacity, the improvement decreases as the slot capacity increases. On the contrary, the packet loss probability increases as the slot capacity increases. In addition to the mean value analysis, the delay distribution and the 95% delay of data packets are provided.

This paper presents a prototype of a distributed 3D rendering system in a hierarchical Grid environment. 3D rendering with massive data sets is a computationally intensive task. In order to make full use of computational resources on Grids, a hierarchical system architecture is designed to run over multiple clusters. This architecture involves both sort-first and sort-last parallel rendering algorithms to achieve excellent scalability, rendering performance and load balance.

A new inter-client synchronization framework employing a server-client coordinated adaptive playout and error control toward one-to-many (i.e., multicast) media streaming is discussed in this paper. The proposed adaptive playout mechanism controls the playout speed of audio and video by adopting the time-scale modification of audio. Based on the overall synchronization status as well as the buffer occupancy level, the playout speed of each client is manipulated within a perceptually tolerable range. By coordinating the playout speed of each client, the inter-client synchronization with respect to the target presentation time is smoothly achieved. Furthermore, RTCP-compatible signaling between the server and group-clients is performed to achieve the inter-client synchronization and error recovery, where the exchange of controlling message is restricted. Simulation results show the performance of the proposed multicast media streaming framework.

This letter shows the performance comparisons of several different rate scheduling schemes for non-real time data service over the uplink of burst switching-based direct sequence code division multiple access (DS/CDMA) system to support the integrated voice/data service. The closed-form solution of optimal scheduling formulation, which minimizes average transmission delay when all of the active data users are transmitting simultaneously, is presented and mathematical analyses with other rate scheduling schemes, which provide efficiency criterion of transmission delay for rate scheduling schemes, are performed. Numerical results show the analyses explicitly.

We analyze the Lagarias-Odlyzko low-density attack precisely, and show that this low-density attack can be applied to the Chor-Rivest and the Okamoto-Tanaka-Uchiyama cryptosystemes, which are considered to be secure against the low-density attack. According to our analysis, these schemes turn out to be no longer secure against the low-density attack.

This letter deals with the common randomness problem formulated by Ahlswede and Csiszar. Especially, we consider their source-type models without wiretapper for ergodic sources, and clarify the secret key-capacity by using the bin coding technique proposed by Cover.